git.sesse.net Git - vlc/blob - modules/audio_filter/channel_mixer/mono.c

   1 /*****************************************************************************
   2  * mono.c : stereo2mono downmixsimple channel mixer plug-in
   3  *****************************************************************************
   4  * Copyright (C) 2006 M2X
   5  * $Id$
   6  *
   7  * Authors: Jean-Paul Saman <jpsaman at m2x dot nl>
   8  *
   9  * This program is free software; you can redistribute it and/or modify
  10  * it under the terms of the GNU General Public License as published by
  11  * the Free Software Foundation; either version 2 of the License, or
  12  * (at your option) any later version.
  13  *
  14  * This program is distributed in the hope that it will be useful,
  15  * but WITHOUT ANY WARRANTY; without even the implied warranty of
  16  * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
  17  * GNU General Public License for more details.
  18  *
  19  * You should have received a copy of the GNU General Public License
  20  * along with this program; if not, write to the Free Software
  21  * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston MA 02110-1301, USA.
  22  *****************************************************************************/
  23
  24 /*****************************************************************************
  25  * Preamble
  26  *****************************************************************************/
  27 #include <string.h>
  28 #include <math.h>                                        /* sqrt */
  29
  30 #ifdef HAVE_STDINT_H
  31 #   include <stdint.h>                                         /* int16_t .. */
  32 #elif defined(HAVE_INTTYPES_H)
  33 #   include <inttypes.h>                                       /* int16_t .. */
  34 #endif
  35
  36 #ifdef HAVE_UNISTD_H
  37 #   include <unistd.h>
  38 #endif
  39
  40 #include <vlc/vlc.h>
  41 #include <vlc_es.h>
  42 #include <vlc_block.h>
  43 #include <vlc_filter.h>
  44 #include <vlc_aout.h>
  45
  46 /*****************************************************************************
  47  * Local prototypes
  48  *****************************************************************************/
  49 static int  OpenFilter    ( vlc_object_t * );
  50 static void CloseFilter   ( vlc_object_t * );
  51
  52 static block_t *Convert( filter_t *p_filter, block_t *p_block );
  53
  54 static unsigned int stereo_to_mono( aout_instance_t *, aout_filter_t *,
  55                                     aout_buffer_t *, aout_buffer_t * );
  56 static unsigned int mono( aout_instance_t *, aout_filter_t *,
  57                                     aout_buffer_t *, aout_buffer_t * );
  58 static void stereo2mono_downmix( aout_instance_t *, aout_filter_t *,
  59                                  aout_buffer_t *, aout_buffer_t * );
  60
  61 /*****************************************************************************
  62  * Local structures
  63  *****************************************************************************/
  64 struct atomic_operation_t
  65 {
  66     int i_source_channel_offset;
  67     int i_dest_channel_offset;
  68     unsigned int i_delay;/* in sample unit */
  69     double d_amplitude_factor;
  70 };
  71
  72 struct filter_sys_t
  73 {
  74     vlc_bool_t b_downmix;
  75
  76     unsigned int i_nb_channels; /* number of int16_t per sample */
  77     int i_channel_selected;
  78     int i_bitspersample;
  79
  80     size_t i_overflow_buffer_size;/* in bytes */
  81     byte_t * p_overflow_buffer;
  82     unsigned int i_nb_atomic_operations;
  83     struct atomic_operation_t * p_atomic_operations;
  84 };
  85
  86 #define MONO_DOWNMIX_TEXT N_("Use downmix algorithme.")
  87 #define MONO_DOWNMIX_LONGTEXT N_("This option selects a stereo to mono " \
  88     "downmix algorithm that is used in the headphone channel mixer. It" \
  89     "gives the effect of standing in a room full of speakers." )
  90
  91 #define MONO_CHANNEL_TEXT N_("Select channel to keep")
  92 #define MONO_CHANNEL_LONGTEXT N_("This option silences all other channels " \
  93     "except the selected channel. Choose one from (0=left, 1=right " \
  94     "2=rear left, 3=rear right, 4=center, 5=left front)")
  95
  96 static const int pi_pos_values[] = { 0, 1, 2, 4, 8, 5 };
  97 static const char *ppsz_pos_descriptions[] =
  98 { N_("Left"), N_("Right"), N_("Left rear"), N_("Right rear"), N_("Center"),
  99   N_("Left front") };
 100
 101 /* our internal channel order (WG-4 order) */
 102 static const uint32_t pi_channels_out[] =
 103 { AOUT_CHAN_LEFT, AOUT_CHAN_RIGHT, AOUT_CHAN_REARLEFT, AOUT_CHAN_REARRIGHT,
 104   AOUT_CHAN_CENTER, AOUT_CHAN_LFE, 0 };
 105
 106 #define MONO_CFG "sout-mono-"
 107 /*****************************************************************************
 108  * Module descriptor
 109  *****************************************************************************/
 110 vlc_module_begin();
 111     set_description( _("Audio filter for stereo to mono conversion") );
 112     set_capability( "audio filter2", 0 );
 113
 114     add_bool( MONO_CFG "downmix", VLC_FALSE, NULL, MONO_DOWNMIX_TEXT,
 115               MONO_DOWNMIX_LONGTEXT, VLC_FALSE );
 116     add_integer( MONO_CFG "channel", -1, NULL, MONO_CHANNEL_TEXT,
 117         MONO_CHANNEL_LONGTEXT, VLC_FALSE );
 118         change_integer_list( pi_pos_values, ppsz_pos_descriptions, 0 );
 119
 120     set_category( CAT_AUDIO );
 121     set_subcategory( SUBCAT_AUDIO_MISC );
 122     set_callbacks( OpenFilter, CloseFilter );
 123     set_shortname( "Mono" );
 124 vlc_module_end();
 125
 126 /* Init() and ComputeChannelOperations() -
 127  * Code taken from modules/audio_filter/channel_mixer/headphone.c
 128  * converted from float into int16_t based downmix
 129  * Written by Boris Dorès <babal@via.ecp.fr>
 130  */
 131
 132 /*****************************************************************************
 133  * Init: initialize internal data structures
 134  * and computes the needed atomic operations
 135  *****************************************************************************/
 136 /* x and z represent the coordinates of the virtual speaker
 137  *  relatively to the center of the listener's head, measured in meters :
 138  *
 139  *  left              right
 140  *Z
 141  *-
 142  *a          head
 143  *x
 144  *i
 145  *s
 146  *  rear left    rear right
 147  *
 148  *          x-axis
 149  *  */
 150 static void ComputeChannelOperations( struct filter_sys_t * p_data,
 151         unsigned int i_rate, unsigned int i_next_atomic_operation,
 152         int i_source_channel_offset, double d_x, double d_z,
 153         double d_compensation_length, double d_channel_amplitude_factor )
 154 {
 155     double d_c = 340; /*sound celerity (unit: m/s)*/
 156     double d_compensation_delay = (d_compensation_length-0.1) / d_c * i_rate;
 157
 158     /* Left ear */
 159     p_data->p_atomic_operations[i_next_atomic_operation]
 160         .i_source_channel_offset = i_source_channel_offset;
 161     p_data->p_atomic_operations[i_next_atomic_operation]
 162         .i_dest_channel_offset = 0;/* left */
 163     p_data->p_atomic_operations[i_next_atomic_operation]
 164         .i_delay = (int)( sqrt( (-0.1-d_x)*(-0.1-d_x) + (0-d_z)*(0-d_z) )
 165                           / d_c * i_rate - d_compensation_delay );
 166     if( d_x < 0 )
 167     {
 168         p_data->p_atomic_operations[i_next_atomic_operation]
 169             .d_amplitude_factor = d_channel_amplitude_factor * 1.1 / 2;
 170     }
 171     else if( d_x > 0 )
 172     {
 173         p_data->p_atomic_operations[i_next_atomic_operation]
 174             .d_amplitude_factor = d_channel_amplitude_factor * 0.9 / 2;
 175     }
 176     else
 177     {
 178         p_data->p_atomic_operations[i_next_atomic_operation]
 179             .d_amplitude_factor = d_channel_amplitude_factor / 2;
 180     }
 181
 182     /* Right ear */
 183     p_data->p_atomic_operations[i_next_atomic_operation + 1]
 184         .i_source_channel_offset = i_source_channel_offset;
 185     p_data->p_atomic_operations[i_next_atomic_operation + 1]
 186         .i_dest_channel_offset = 1;/* right */
 187     p_data->p_atomic_operations[i_next_atomic_operation + 1]
 188         .i_delay = (int)( sqrt( (0.1-d_x)*(0.1-d_x) + (0-d_z)*(0-d_z) )
 189                           / d_c * i_rate - d_compensation_delay );
 190     if( d_x < 0 )
 191     {
 192         p_data->p_atomic_operations[i_next_atomic_operation + 1]
 193             .d_amplitude_factor = d_channel_amplitude_factor * 0.9 / 2;
 194     }
 195     else if( d_x > 0 )
 196     {
 197         p_data->p_atomic_operations[i_next_atomic_operation + 1]
 198             .d_amplitude_factor = d_channel_amplitude_factor * 1.1 / 2;
 199     }
 200     else
 201     {
 202         p_data->p_atomic_operations[i_next_atomic_operation + 1]
 203             .d_amplitude_factor = d_channel_amplitude_factor / 2;
 204     }
 205 }
 206
 207 static int Init( vlc_object_t *p_this, struct filter_sys_t * p_data,
 208                  unsigned int i_nb_channels, uint32_t i_physical_channels,
 209                  unsigned int i_rate )
 210 {
 211     double d_x = config_GetInt( p_this, "headphone-dim" );
 212     double d_z = d_x;
 213     double d_z_rear = -d_x/3;
 214     double d_min = 0;
 215     unsigned int i_next_atomic_operation;
 216     int i_source_channel_offset;
 217     unsigned int i;
 218
 219     if( p_data == NULL )
 220     {
 221         msg_Dbg( p_this, "passing a null pointer as argument" );
 222         return 0;
 223     }
 224
 225     if( config_GetInt( p_this, "headphone-compensate" ) )
 226     {
 227         /* minimal distance to any speaker */
 228         if( i_physical_channels & AOUT_CHAN_REARCENTER )
 229         {
 230             d_min = d_z_rear;
 231         }
 232         else
 233         {
 234             d_min = d_z;
 235         }
 236     }
 237
 238     /* Number of elementary operations */
 239     p_data->i_nb_atomic_operations = i_nb_channels * 2;
 240     if( i_physical_channels & AOUT_CHAN_CENTER )
 241     {
 242         p_data->i_nb_atomic_operations += 2;
 243     }
 244     p_data->p_atomic_operations = malloc( sizeof(struct atomic_operation_t)
 245             * p_data->i_nb_atomic_operations );
 246     if( p_data->p_atomic_operations == NULL )
 247     {
 248         msg_Err( p_this, "out of memory" );
 249         return -1;
 250     }
 251
 252     /* For each virtual speaker, computes elementary wave propagation time
 253      * to each ear */
 254     i_next_atomic_operation = 0;
 255     i_source_channel_offset = 0;
 256     if( i_physical_channels & AOUT_CHAN_LEFT )
 257     {
 258         ComputeChannelOperations( p_data , i_rate
 259                 , i_next_atomic_operation , i_source_channel_offset
 260                 , -d_x , d_z , d_min , 2.0 / i_nb_channels );
 261         i_next_atomic_operation += 2;
 262         i_source_channel_offset++;
 263     }
 264     if( i_physical_channels & AOUT_CHAN_RIGHT )
 265     {
 266         ComputeChannelOperations( p_data , i_rate
 267                 , i_next_atomic_operation , i_source_channel_offset
 268                 , d_x , d_z , d_min , 2.0 / i_nb_channels );
 269         i_next_atomic_operation += 2;
 270         i_source_channel_offset++;
 271     }
 272     if( i_physical_channels & AOUT_CHAN_MIDDLELEFT )
 273     {
 274         ComputeChannelOperations( p_data , i_rate
 275                 , i_next_atomic_operation , i_source_channel_offset
 276                 , -d_x , 0 , d_min , 1.5 / i_nb_channels );
 277         i_next_atomic_operation += 2;
 278         i_source_channel_offset++;
 279     }
 280     if( i_physical_channels & AOUT_CHAN_MIDDLERIGHT )
 281     {
 282         ComputeChannelOperations( p_data , i_rate
 283                 , i_next_atomic_operation , i_source_channel_offset
 284                 , d_x , 0 , d_min , 1.5 / i_nb_channels );
 285         i_next_atomic_operation += 2;
 286         i_source_channel_offset++;
 287     }
 288     if( i_physical_channels & AOUT_CHAN_REARLEFT )
 289     {
 290         ComputeChannelOperations( p_data , i_rate
 291                 , i_next_atomic_operation , i_source_channel_offset
 292                 , -d_x , d_z_rear , d_min , 1.5 / i_nb_channels );
 293         i_next_atomic_operation += 2;
 294         i_source_channel_offset++;
 295     }
 296     if( i_physical_channels & AOUT_CHAN_REARRIGHT )
 297     {
 298         ComputeChannelOperations( p_data , i_rate
 299                 , i_next_atomic_operation , i_source_channel_offset
 300                 , d_x , d_z_rear , d_min , 1.5 / i_nb_channels );
 301         i_next_atomic_operation += 2;
 302         i_source_channel_offset++;
 303     }
 304     if( i_physical_channels & AOUT_CHAN_REARCENTER )
 305     {
 306         ComputeChannelOperations( p_data , i_rate
 307                 , i_next_atomic_operation , i_source_channel_offset
 308                 , 0 , -d_z , d_min , 1.5 / i_nb_channels );
 309         i_next_atomic_operation += 2;
 310         i_source_channel_offset++;
 311     }
 312     if( i_physical_channels & AOUT_CHAN_CENTER )
 313     {
 314         /* having two center channels increases the spatialization effect */
 315         ComputeChannelOperations( p_data , i_rate
 316                 , i_next_atomic_operation , i_source_channel_offset
 317                 , d_x / 5.0 , d_z , d_min , 0.75 / i_nb_channels );
 318         i_next_atomic_operation += 2;
 319         ComputeChannelOperations( p_data , i_rate
 320                 , i_next_atomic_operation , i_source_channel_offset
 321                 , -d_x / 5.0 , d_z , d_min , 0.75 / i_nb_channels );
 322         i_next_atomic_operation += 2;
 323         i_source_channel_offset++;
 324     }
 325     if( i_physical_channels & AOUT_CHAN_LFE )
 326     {
 327         ComputeChannelOperations( p_data , i_rate
 328                 , i_next_atomic_operation , i_source_channel_offset
 329                 , 0 , d_z_rear , d_min , 5.0 / i_nb_channels );
 330         i_next_atomic_operation += 2;
 331         i_source_channel_offset++;
 332     }
 333
 334     /* Initialize the overflow buffer
 335      * we need it because the process induce a delay in the samples */
 336     p_data->i_overflow_buffer_size = 0;
 337     for( i = 0 ; i < p_data->i_nb_atomic_operations ; i++ )
 338     {
 339         if( p_data->i_overflow_buffer_size
 340                 < p_data->p_atomic_operations[i].i_delay * 2 * sizeof (int16_t) )
 341         {
 342             p_data->i_overflow_buffer_size
 343                 = p_data->p_atomic_operations[i].i_delay * 2 * sizeof (int16_t);
 344         }
 345     }
 346     p_data->p_overflow_buffer = malloc( p_data->i_overflow_buffer_size );
 347     if( p_data->p_atomic_operations == NULL )
 348     {
 349         msg_Err( p_this, "out of memory" );
 350         return -1;
 351     }
 352     memset( p_data->p_overflow_buffer, 0, p_data->i_overflow_buffer_size );
 353
 354     /* end */
 355     return 0;
 356 }
 357
 358 /*****************************************************************************
 359  * OpenFilter
 360  *****************************************************************************/
 361 static int OpenFilter( vlc_object_t *p_this )
 362 {
 363     filter_t * p_filter = (filter_t *)p_this;
 364     filter_sys_t *p_sys = NULL;
 365
 366     if( aout_FormatNbChannels( &(p_filter->fmt_in.audio) ) == 1 )
 367     {
 368         msg_Dbg( p_filter, "filter discarded (incompatible format)" );
 369         return VLC_EGENERIC;
 370     }
 371
 372     if( (p_filter->fmt_in.i_codec != AOUT_FMT_S16_NE) ||
 373         (p_filter->fmt_out.i_codec != AOUT_FMT_S16_NE) )
 374     {
 375         msg_Err( p_this, "filter discarded (invalid format)" );
 376         return -1;
 377     }
 378
 379     if( (p_filter->fmt_in.audio.i_format != p_filter->fmt_out.audio.i_format) &&
 380         (p_filter->fmt_in.audio.i_rate != p_filter->fmt_out.audio.i_rate) &&
 381         (p_filter->fmt_in.audio.i_format != AOUT_FMT_S16_NE) &&
 382         (p_filter->fmt_out.audio.i_format != AOUT_FMT_S16_NE) &&
 383         (p_filter->fmt_in.audio.i_bitspersample !=
 384                                     p_filter->fmt_out.audio.i_bitspersample))
 385     {
 386         msg_Err( p_this, "couldn't load mono filter" );
 387         return VLC_EGENERIC;
 388     }
 389
 390     /* Allocate the memory needed to store the module's structure */
 391     p_sys = p_filter->p_sys = malloc( sizeof(filter_sys_t) );
 392     if( p_sys == NULL )
 393     {
 394         msg_Err( p_filter, "out of memory" );
 395         return VLC_EGENERIC;
 396     }
 397
 398     var_Create( p_this, MONO_CFG "downmix",
 399                 VLC_VAR_BOOL | VLC_VAR_DOINHERIT );
 400     p_sys->b_downmix = var_GetBool( p_this, MONO_CFG "downmix" );
 401
 402     var_Create( p_this, MONO_CFG "channel",
 403                 VLC_VAR_INTEGER | VLC_VAR_DOINHERIT );
 404     p_sys->i_channel_selected =
 405             (unsigned int) var_GetInteger( p_this, MONO_CFG "channel" );
 406
 407     if( p_sys->b_downmix )
 408     {
 409         msg_Dbg( p_this, "using stereo to mono downmix" );
 410         p_filter->fmt_out.audio.i_physical_channels = AOUT_CHAN_CENTER;
 411         p_filter->fmt_out.audio.i_channels = 1;
 412     }
 413     else
 414     {
 415         msg_Dbg( p_this, "using pseudo mono" );
 416         p_filter->fmt_out.audio.i_physical_channels =
 417                             (AOUT_CHAN_LEFT | AOUT_CHAN_RIGHT);
 418         p_filter->fmt_out.audio.i_channels = 2;
 419     }
 420
 421     p_filter->fmt_out.audio.i_rate = p_filter->fmt_in.audio.i_rate;
 422     p_filter->fmt_out.audio.i_format = p_filter->fmt_out.i_codec;
 423
 424     p_sys->i_nb_channels = aout_FormatNbChannels( &(p_filter->fmt_in.audio) );
 425     p_sys->i_bitspersample = p_filter->fmt_out.audio.i_bitspersample;
 426
 427     p_sys->i_overflow_buffer_size = 0;
 428     p_sys->p_overflow_buffer = NULL;
 429     p_sys->i_nb_atomic_operations = 0;
 430     p_sys->p_atomic_operations = NULL;
 431
 432     if( Init( VLC_OBJECT(p_filter), p_filter->p_sys,
 433               aout_FormatNbChannels( &p_filter->fmt_in.audio ),
 434               p_filter->fmt_in.audio.i_physical_channels,
 435               p_filter->fmt_in.audio.i_rate ) < 0 )
 436     {
 437         return VLC_EGENERIC;
 438     }
 439
 440     p_filter->pf_audio_filter = Convert;
 441
 442     msg_Dbg( p_this, "%4.4s->%4.4s, channels %d->%d, bits per sample: %i->%i",
 443              (char *)&p_filter->fmt_in.i_codec,
 444              (char *)&p_filter->fmt_out.i_codec,
 445              p_filter->fmt_in.audio.i_physical_channels,
 446              p_filter->fmt_out.audio.i_physical_channels,
 447              p_filter->fmt_in.audio.i_bitspersample,
 448              p_filter->fmt_out.audio.i_bitspersample );
 449
 450     return VLC_SUCCESS;
 451 }
 452
 453 /*****************************************************************************
 454  * CloseFilter
 455  *****************************************************************************/
 456 static void CloseFilter( vlc_object_t *p_this)
 457 {
 458     filter_t *p_filter = (filter_t *) p_this;
 459     filter_sys_t *p_sys = p_filter->p_sys;
 460
 461     var_Destroy( p_this, MONO_CFG "channel" );
 462     var_Destroy( p_this, MONO_CFG "downmix" );
 463     free( p_sys );
 464 }
 465
 466 /*****************************************************************************
 467  * Convert
 468  *****************************************************************************/
 469 static block_t *Convert( filter_t *p_filter, block_t *p_block )
 470 {
 471     aout_filter_t aout_filter;
 472     aout_buffer_t in_buf, out_buf;
 473     block_t *p_out = NULL;
 474     unsigned int i_samples;
 475     int i_out_size;
 476
 477     if( !p_block || !p_block->i_samples )
 478     {
 479         if( p_block )
 480             p_block->pf_release( p_block );
 481         return NULL;
 482     }
 483
 484     i_out_size = p_block->i_samples * p_filter->p_sys->i_bitspersample/8 *
 485                  aout_FormatNbChannels( &(p_filter->fmt_out.audio) );
 486
 487     p_out = p_filter->pf_audio_buffer_new( p_filter, i_out_size );
 488     if( !p_out )
 489     {
 490         msg_Warn( p_filter, "can't get output buffer" );
 491         p_block->pf_release( p_block );
 492         return NULL;
 493     }
 494     p_out->i_samples = (p_block->i_samples / p_filter->p_sys->i_nb_channels) *
 495                        aout_FormatNbChannels( &(p_filter->fmt_out.audio) );
 496     p_out->i_dts = p_block->i_dts;
 497     p_out->i_pts = p_block->i_pts;
 498     p_out->i_length = p_block->i_length;
 499
 500     aout_filter.p_sys = (struct aout_filter_sys_t *)p_filter->p_sys;
 501     aout_filter.input = p_filter->fmt_in.audio;
 502     aout_filter.input.i_format = p_filter->fmt_in.i_codec;
 503     aout_filter.output = p_filter->fmt_out.audio;
 504     aout_filter.output.i_format = p_filter->fmt_out.i_codec;
 505
 506     in_buf.p_buffer = p_block->p_buffer;
 507     in_buf.i_nb_bytes = p_block->i_buffer;
 508     in_buf.i_nb_samples = p_block->i_samples;
 509
 510 #if 0
 511     unsigned int i_in_size = in_buf.i_nb_samples  * (p_filter->p_sys->i_bitspersample/8) *
 512                              aout_FormatNbChannels( &(p_filter->fmt_in.audio) );
 513     if( (in_buf.i_nb_bytes != i_in_size) && ((i_in_size % 32) != 0) ) /* is it word aligned?? */
 514     {
 515         msg_Err( p_filter, "input buffer is not word aligned" );
 516         /* Fix output buffer to be word aligned */
 517     }
 518 #endif
 519
 520     out_buf.p_buffer = p_out->p_buffer;
 521     out_buf.i_nb_bytes = p_out->i_buffer;
 522     out_buf.i_nb_samples = p_out->i_samples;
 523
 524     memset( p_out->p_buffer, 0, i_out_size );
 525     if( p_filter->p_sys->b_downmix )
 526     {
 527         stereo2mono_downmix( (aout_instance_t *)p_filter, &aout_filter,
 528                              &in_buf, &out_buf );
 529         i_samples = mono( (aout_instance_t *)p_filter, &aout_filter,
 530                            &out_buf, &in_buf );
 531     }
 532     else
 533     {
 534         i_samples = stereo_to_mono( (aout_instance_t *)p_filter, &aout_filter,
 535                                     &out_buf, &in_buf );
 536     }
 537
 538     p_out->i_buffer = out_buf.i_nb_bytes;
 539     p_out->i_samples = out_buf.i_nb_samples;
 540
 541     p_block->pf_release( p_block );
 542     return p_out;
 543 }
 544
 545 /* stereo2mono_downmix - stereo channels into one mono channel.
 546  * Code taken from modules/audio_filter/channel_mixer/headphone.c
 547  * converted from float into int16_t based downmix
 548  * Written by Boris Dorès <babal@via.ecp.fr>
 549  */
 550 static void stereo2mono_downmix( aout_instance_t * p_aout, aout_filter_t * p_filter,
 551                             aout_buffer_t * p_in_buf, aout_buffer_t * p_out_buf )
 552 {
 553     filter_sys_t *p_sys = (filter_sys_t *)p_filter->p_sys;
 554
 555     int i_input_nb = aout_FormatNbChannels( &p_filter->input );
 556     int i_output_nb = aout_FormatNbChannels( &p_filter->output );
 557
 558     int16_t * p_in = (int16_t*) p_in_buf->p_buffer;
 559     byte_t * p_out;
 560     byte_t * p_overflow;
 561     byte_t * p_slide;
 562
 563     size_t i_overflow_size;     /* in bytes */
 564     size_t i_out_size;          /* in bytes */
 565
 566     unsigned int i, j;
 567
 568     int i_source_channel_offset;
 569     int i_dest_channel_offset;
 570     unsigned int i_delay;
 571     double d_amplitude_factor;
 572
 573     /* out buffer characterisitcs */
 574     p_out_buf->i_nb_samples = p_in_buf->i_nb_samples;
 575     p_out_buf->i_nb_bytes = p_in_buf->i_nb_bytes * i_output_nb / i_input_nb;
 576     p_out = p_out_buf->p_buffer;
 577     i_out_size = p_out_buf->i_nb_bytes;
 578
 579     if( p_sys != NULL )
 580     {
 581         /* Slide the overflow buffer */
 582         p_overflow = p_sys->p_overflow_buffer;
 583         i_overflow_size = p_sys->i_overflow_buffer_size;
 584
 585         if ( i_out_size > i_overflow_size )
 586             memcpy( p_out, p_overflow, i_overflow_size );
 587         else
 588             memcpy( p_out, p_overflow, i_out_size );
 589
 590         p_slide = p_sys->p_overflow_buffer;
 591         while( p_slide < p_overflow + i_overflow_size )
 592         {
 593             if( p_slide + i_out_size < p_overflow + i_overflow_size )
 594             {
 595                 memset( p_slide, 0, i_out_size );
 596                 if( p_slide + 2 * i_out_size < p_overflow + i_overflow_size )
 597                     memcpy( p_slide, p_slide + i_out_size, i_out_size );
 598                 else
 599                     memcpy( p_slide, p_slide + i_out_size,
 600                             p_overflow + i_overflow_size - ( p_slide + i_out_size ) );
 601             }
 602             else
 603             {
 604                 memset( p_slide, 0, p_overflow + i_overflow_size - p_slide );
 605             }
 606             p_slide += i_out_size;
 607         }
 608
 609         /* apply the atomic operations */
 610         for( i = 0; i < p_sys->i_nb_atomic_operations; i++ )
 611         {
 612             /* shorter variable names */
 613             i_source_channel_offset
 614                 = p_sys->p_atomic_operations[i].i_source_channel_offset;
 615             i_dest_channel_offset
 616                 = p_sys->p_atomic_operations[i].i_dest_channel_offset;
 617             i_delay = p_sys->p_atomic_operations[i].i_delay;
 618             d_amplitude_factor
 619                 = p_sys->p_atomic_operations[i].d_amplitude_factor;
 620
 621             if( p_out_buf->i_nb_samples > i_delay )
 622             {
 623                 /* current buffer coefficients */
 624                 for( j = 0; j < p_out_buf->i_nb_samples - i_delay; j++ )
 625                 {
 626                     ((int16_t*)p_out)[ (i_delay+j)*i_output_nb + i_dest_channel_offset ]
 627                         += p_in[ j * i_input_nb + i_source_channel_offset ]
 628                            * d_amplitude_factor;
 629                 }
 630
 631                 /* overflow buffer coefficients */
 632                 for( j = 0; j < i_delay; j++ )
 633                 {
 634                     ((int16_t*)p_overflow)[ j*i_output_nb + i_dest_channel_offset ]
 635                         += p_in[ (p_out_buf->i_nb_samples - i_delay + j)
 636                            * i_input_nb + i_source_channel_offset ]
 637                            * d_amplitude_factor;
 638                 }
 639             }
 640             else
 641             {
 642                 /* overflow buffer coefficients only */
 643                 for( j = 0; j < p_out_buf->i_nb_samples; j++ )
 644                 {
 645                     ((int16_t*)p_overflow)[ (i_delay - p_out_buf->i_nb_samples + j)
 646                         * i_output_nb + i_dest_channel_offset ]
 647                         += p_in[ j * i_input_nb + i_source_channel_offset ]
 648                            * d_amplitude_factor;
 649                 }
 650             }
 651         }
 652     }
 653     else
 654     {
 655         memset( p_out, 0, i_out_size );
 656     }
 657 }
 658
 659 /* Simple stereo to mono mixing. */
 660 static unsigned int mono( aout_instance_t * p_aout, aout_filter_t *p_filter,
 661                           aout_buffer_t *p_output, aout_buffer_t *p_input )
 662 {
 663     filter_sys_t *p_sys = (filter_sys_t *)p_filter->p_sys;
 664     int16_t *p_in, *p_out;
 665     unsigned int n = 0, r = 0;
 666
 667     p_in = (int16_t *) p_input->p_buffer;
 668     p_out = (int16_t *) p_output->p_buffer;
 669
 670     while( n < (p_input->i_nb_samples * p_sys->i_nb_channels) )
 671     {
 672         p_out[r] = (p_in[n] + p_in[n+1]) >> 1;
 673         r++;
 674         n += 2;
 675     }
 676     return r;
 677 }
 678
 679 /* Simple stereo to mono mixing. */
 680 static unsigned int stereo_to_mono( aout_instance_t * p_aout, aout_filter_t *p_filter,
 681                                     aout_buffer_t *p_output, aout_buffer_t *p_input )
 682 {
 683     filter_sys_t *p_sys = (filter_sys_t *)p_filter->p_sys;
 684     int16_t *p_in, *p_out;
 685     unsigned int n;
 686
 687     p_in = (int16_t *) p_input->p_buffer;
 688     p_out = (int16_t *) p_output->p_buffer;
 689
 690     for( n = 0; n < (p_input->i_nb_samples * p_sys->i_nb_channels); n++ )
 691     {
 692         /* Fake real mono. */
 693         if( p_sys->i_channel_selected == -1)
 694         {
 695             p_out[n] = p_out[n+1] = (p_in[n] + p_in[n+1]) >> 1;
 696             n++;
 697         }
 698         else if( (n % p_sys->i_nb_channels) == (unsigned int) p_sys->i_channel_selected )
 699         {
 700             p_out[n] = p_out[n+1] = p_in[n];
 701         }
 702     }
 703     return n;
 704 }